Network telephony appliance and system for inter/intranet telephony

ABSTRACT

A network appliance ( 100 ) is provided having a network controller subsystem ( 110 ) for coupling the appliance ( 100 ) to a data network for providing and receiving data packets to and from a packet data network. A digital signal processing subsystem ( 120 ) is coupled to the network controller subsystem ( 110 ). A signal conversion subsystem ( 130 ) is coupled to the digital signal processing subsystem ( 120 ) and a user interface subsystem ( 160 ) is coupled to both the signal conversion subsystem ( 130 ) and the digital signal processing subsystem ( 120 ). The digital signal processing subsystem ( 120 ) operates under the control of a computer program which is capable of detecting incoming calls, initiating call sessions, and preferably, implementing advanced telephony features.

FIELD OF THE INVENTION

The present invention relates in general to the field of Internet andintranet telephony. More particularly, the present invention relates toa network telecommunications appliance and system for Internet/intranetcommunications.

BACKGROUND OF THE INVENTION

Over recent years, the Internet has evolved from a convenient additionalmeans of communications to an essential communication tool in thebusiness, technical and educational fields. In this regard, a growingsegment of the Internet relates to Internet telephony which provides anumber of advantages over conventional circuit-switched networkcontrolled by a separate signaling network. For one thing, parties areallowed to more easily select and use encoding and other datacompression techniques that are most appropriate for their qualityneeds. Parties may, for example, decide that for international calls,they would trade lower cost for full toll quality, while a reportercalling in her story to a radio station may go for full FM quality withlittle regard for price. Even without quality degradation. 5.3 kb/s(G.723.1) to 8 kb/s (G.729) are sufficient to support close to tollquality as opposed to 64 kb/s for conventional landline telephonenetworks. This flexibility also has the advantage that during severenetwork overload, e.g., after a natural catastrophe, telephone customerscan still communicate at about 3 kb/s, thus increasing network capacitytwenty-fold.

While it is logical to extend telephony services to existing datanetworks, such as the Internet, because of the intelligence required inthe end systems, cost poses a major disadvantage. Previously, it hasbeen difficult to build packet voice “telephones” requiring no externalpower that operate over low-grade twisted pair wires several miles longat the cost of a basic analog phone.

In addition, the majority of known Internet telephony products aredesigned to operate in accordance with the H.323 signaling and controlprotocol. The H.323 protocol is a complex protocol which is difficult touse and implement. As a result of this complexity, differentimplementations of H.323 devices may be adversely affected bycompatibility issues. In addition, devices operating under the H.323protocol cannot communicate directly with one another, calls must beprocessed and routed by a telephony server.

According, there remains a need for a network telephony appliance whichis low cost, operates using a simple signaling protocol and offers avast set of advanced telephony features.

SUMMARY OF THE INVENTION

The afore described limitations and inadequacies of conventionaltelephone systems and known Internet telephony systems are substantiallyovercome by the present invention, in which a primary object is toprovide a packet-based voice communication system for use over theInternet and intranet telecommunications networks.

It is another object of the present invention to provide a packet datatelephony appliance for use over a data network, such as an Ethernetnetwork.

It is still another object of the present invention to provide acommunication protocol for use in a packet-based telecommunicationsystem.

It is yet another object of the present invention to provide an Internetprotocol architecture which supports telephony and othercontinuous-media or streaming media services such as “Internet radio”and “Internet TV.”

It is yet another object of the present invention to provide a low cost,stand alone network telephony appliance capable of direct calling ofanother network telephone station or indirectly calling another networktelephone station party, such as through a redirect server.

In accordance with a first embodiment of the present invention, anetwork packet data telephone apparatus is provided that includes: anetwork controller, such as an Ethernet controller subsystem, coupled toa data network for providing and receiving data packets to and from thenetwork. A digital signal processing subsystem is coupled to the networkcontroller subsystem and operates under a computer program for detectingincoming calls, initiating call sessions and implementing telephonyfeatures. A signal conversion subsystem is coupled to the digital signalprocessing subsystem for converting digital packet information intoanalog signals and vice versa. A user interface subsystem is coupled toboth the signal conversion subsystem and the digital signal processingsubsystem for providing user control and feedback to the apparatus. Thisstand alone network telephony device is referred to herein as a networktelephony appliance.

Preferably, the computer program of the network telephony applianceimplements the Session Initiation Protocol (SIP). In this case, a uniqueSIP address is associated with the device and session initiation andcontrol are performed in accordance with the SIP protocol.

The network telephony appliance preferably implements high leveltelephony functionality including a monitoring feature, call forwarding,streaming audio mode, caller log, callee log and the like.

Preferably, the network telephony appliance includes sensor interfacecircuitry for receiving signals from remote sources, such as sensors.The signals received from the remote sources are processed by thenetwork telephony appliance and sent to an appropriate networkdestination.

In another aspect of the present invention, a communication protocol isprovided for use in a packet-based telecommunication system, thecommunication protocol having: an Ethernet protocol layer; an InternetProtocol (IP) layer stacked on top of the Ethernet protocol layer forinterfacing with the Ethernet protocol layer; an Address ResolutionProtocol (ARP) layer stacked on top of the Ethernet protocol layer forinterfacing with the Ethernet protocol layer and the IP layer, and fortranslating IP addresses into Media Access Control (MAC) addresses; aUser Datagram Protocol (UDP) layer stacked on top of the ARP and IPlayers for interfacing with the ARP and IP layers and for providingreal-time transport of application data and controls within thetelecommunication system; a Real-Time Transport Protocol (RTP) layerstacked on top of the UDP layer for interfacing with the UDP layer andfor providing real-time audio data transport within thetelecommunication system; one or more control protocol layers stacked ontop of the UDP layer for interfacing with the UDP layer and forsignaling and providing registration of the real-time audio data; andone or more application protocols stacked on top of the RTP layer forinterfacing with the RTP and for formatting the real-time audio data.

In another aspect of the present invention a network telephony systemarchitecture is provided. The system includes at least two networktelephony devices, such as a the present network telephony applianceand/or a general purpose personal computer (PC) with suitable interfacecircuitry and software to operate the PC as a network telephone. Aredirect server is also provided which is coupled to the data networkalong with the network telephony devices. In the system, the networktelephony devices can directly address one another to establish a realtime audio connection. Alternatively, the redirect server can beaccessed by the network telephony devices in order to identify, locate,and initiate a call session with a called party. The redirect server canalso be used to implement high level telephony functions, such as callforwarding, multi-party calling, voice mail and the like.

Further objects, features and advantages of the invention will becomeapparent from the following detailed description taken in conjunctionwith the accompanying figures showing illustrative embodiments of theinvention.

BRIEF DESCRIPTION OF THE DRAWINGS

For a complete understanding of the present invention and the advantagesthereof, reference is now made to the following description taken inconjunction with the accompanying drawings in which like referencenumbers indicate like features and wherein:

FIG. 1 is a illustrative diagram of a telecommunications systemfeaturing a conventional circuit-switched voice network operativelycoupled to a voice packet network;

FIG. 2 is a block diagram of a packet data network telephone system;

FIG. 3 is a diagram showing a protocol stack for telephony devicesoperating on the packet data network telephone system of FIG. 2;

FIG. 4 is a block diagram of a preferred hardware architecture of anetwork telephony appliance in accordance with the present invention;

FIG. 5 is a block diagram further illustrating the network telephonyappliance of FIG. 4;

FIG. 6 is an exemplary memory map for the DSP of the network telephonyappliance of FIG. 5;

FIG. 7 is a block diagram of a memory interface for the DSP of thenetwork telephony appliance of FIG. 5;

FIG. 8 is a block diagram of a network controller interface for the DSPof the network telephony appliance FIG. 5;

FIG. 9 is a block diagram of a codec interface for the DSP of thenetwork telephony appliance of FIG. 5;

FIG. 10 is an exemplary memory map for the DSP of FIG. 5 showing amapping of the LCD control interface to DSP memory addresses;

FIG. 11 is a block diagram showing the software architecture for thenetwork telephony appliance of FIG. 4;

FIG. 12 is a block diagram showing the scheduling mechanisms of theprocess level software of FIG. 11;

FIGS. 13A-F are tables illustrating exemplary task definitions forsoftware operations of a preferred method of operating the Packet datanetwork telephone in accordance with the hardware and softwarearchitectures of FIGS. 4 and 11;

FIG. 14 is a flow diagram of an ARP request output procedure inaccordance with the hardware and software architectures of FIGS. 4, 11and 13;

FIG. 15 is a flow diagram of an ARP request input procedure inaccordance with the hardware and software architectures of FIGS. 4, 11and 13;

FIG. 16 is a diagram showing the IP processing steps in accordance withthe hardware and software architectures of FIGS. 4, 11 and 13;

FIG. 17 is a list of exemplary Ethernet transmit data structuresaccording to the software architecture of FIG. 11;

FIG. 18 is a data flow diagram of a packet sending procedure inaccordance with the hardware and software architectures of FIGS. 4, 11and 13;

FIG. 19 is a data flow diagram of a packet receiving procedure inaccordance with the hardware and software architectures of FIGS. 4, 11and 13;

FIGS. 20A and 20B show the A/D and D/A “ping-pong” buffer scheme used bythe software of the present network telephony appliance;

FIG. 21 is a state transition diagram of the Call_task process of thepresent network telephony appliance;

FIG. 22 is chart defining the key pad values for the preferredembodiment of the Packet data network telephone of FIG. 5;

FIG. 23 is a data structure illustrating key state definitions for thepreferred embodiment of the present network telephony appliance of FIG.5;

FIG. 24 is a mapping of the I/O parallel port of the network telephonyappliance of FIG. 5;

FIG. 25 is a data structure defining the Ethernet controller states ofthe network telephony appliance of FIG. 5;

FIG. 26 is an exemplary RTP header structure for RTP packet processingused in the network telephony appliance network telephony appliance ofFIG. 5;

FIG. 27 is a data structure for use with a tone generation function ofthe Packet data network telephone of FIG. 5;

FIG. 28 is a timing diagram for the tone generation function of thenetwork telephony appliance of FIG. 5;

FIG. 29 is a list of data structures used for processing the SIP_taskrequests or responses in accordance with the network telephony applianceof FIG. 5;

FIG. 30 is a state transition diagram illustrating the network telephonyappliance operating as a client (initiating a call) in accordance withFIG. 5;

FIG. 31 is list of SIP_task responses in accordance with the networktelephony appliance of FIG. 5;

FIG. 32 is a state diagram illustrating the state transition diagram ofa SIP UAS in accordance with the network telephony appliance of FIG. 5;and

FIG. 33 is a block diagram which illustrates part of a packet datanetwork telephony system including one or more network telephonyappliances in accordance with the present invention.

DETAILED DESCRIPTION OF THE INVENTION

FIG. 1 is a block diagram which shows a telecommunications system havingconventional telephony and packet telephony components. As shown in FIG.1, the system includes a circuit-switched voice network 20 coupled to apacket network 30 via a first gateway 12. The figure shows at leastthree possible interactions between Internet telephony and aconventional “plain old telephony service” (POTS) system: “end-to-end”packet delivery; “tail-end hop off” delivery; and local packet delivery.With “end-to-end” packet delivery, end systems such as networkcomputers, dedicated Internet phones or personal computers (PCs) areused to packetize audio and deliver audio packets to one or more similarend systems for playback. With “tail-end hop off” delivery, packetnetworks are used for long-haul voice transmission, while standardcircuit-switched voice circuits are used for connecting customer premiseequipment (CPE), i.e., standard analog telephones, to the packettelephony gateways. “Tail-end hop off” can be used both for individualvoice circuits as well as for PBX interconnects, and allows for thebypassing of conventional long-distance services as well as theinterconnection of POTS equipment to packet-based audio end systems.With local packet delivery, voice data is generated by packet audio endsystems, but carried as circuit-switched voice over leased or publicfacilities.

FIG. 2 shows a preferred embodiment of an packet data network telephonesystem 50 according to the present invention. The packet data networktelephone system includes: an Ethernet LAN 52, Ethernet phones 54, 56,and 58, a workstation 60, a server 62 and a Ethernet gateway 64. TheEthernet phones are network devices, which can take the form of standalone devices, such as a network appliance, or a personal computersystem with audio input and output peripherals and operating under thecontrol of an appropriate computer program. With such an packet datanetwork approach, voice data traffic is packetized proximate the enduser. The packet data network telephony system of FIG. 2, for example,can include several dozen homes, offices or apartments that areconnected to a plurality of Ethernet gateways (only one shown in FIG.2), each of which is located within the CAT-3S cabling distance limit of328 feet from the network termination unit. The gateways can, in turn,connect through optical fiber to the neighborhood switch (not shown), orconnect directly to the Public Switched Telephone Network (PSTN) vialines 66 as shown in FIG. 2. This architecture has the advantage that amix of low-bandwidth and high-bandwidth customers can be accommodatedwithout running additional wires. Since switch costs are dominated byinterface counts rather than bandwidth, this mechanism offers muchhigher per-user bandwidth (particularly peak bandwidth), yet switchingcosts are similar to today's telephone networks. In the architecture ofFIG. 2, each network device includes a network address and each devicecan directly access every other network device via the network address.While a specialized server may be desirable to implement certainfeatures, it is not required to establish a call session, i.e., point topoint data communications between two or more network devices.

The use of a packet data network LAN 52 is advantageous in that it is arelatively inexpensive solution where conventional PC interfaces andnetwork hardware can be used. The Packet data network LAN 52 can beoperated over a variety of media and allows for the easy addition ofmore devices on a multiple-access LAN. Gateway 64 can be a single DSPthat acts as a simple packet voice module and that implements DTMFrecognition for user-to-network signaling.

FIG. 3 is a block diagram which illustrates a packet data networkprotocol stack diagram for providing Internet telephony and othercontinuous-media (“streaming media”) services such as “Internet radio”and “Internet TV.” As known and understood by those skilled in the art,a “protocol” is generally a set of rules for communicating betweencomputers. As such, protocols govern format, timing, sequencing, anderror control. The term “stack” refers to the actual software thatprocesses the protocols and thus allows the use of a specific set orsets of protocols. The diagram shown in FIG. 3 shows how the variousprotocols are interrelated in accordance with the present invention.

The protocol stack 80 of FIG. 3 incorporates a number of layeredprotocols including: a base protocol 82 for providing basic Ethernetmessage format and timing information; an Address Resolution Protocol(ARP) 84 for interfacing with the base protocol 82 and for translatingIP addresses into Media Access Control (MAC) addresses; an InternetProtocol (IP) network layer 86 for interfacing with the base protocol82; a optional Dynamic Host Configuration Protocol (DHCP) 88 forinterfacing with the base protocol 82; and a User Datagram Protocol(UDP) 90 for interfacing with the ARP 84, IP 86 and DHCP 88 protocolsand for real-time transport of application data and controls. Theprotocol stack 80 further includes the following application-specificprotocols for coding speech information: a Real-Time Transport Protocol(RTP) protocol 92 for real-time audio data transport, wherein the RTPprotocol 92 generally interfaces with the UDP 90 and modulation, speechcodec and control applications 94, 96 and 98, respectively. Theapplication protocols 94 and 96 can take several forms, such as theG.711 pulse code modulation and the G.723 speech codec protocols,respectively. In addition, the Real Time Streaming Protocol (RTSP) layer97 can be included to provide enhanced performance in streaming mediaapplications. Control protocol 98 is used for session initiation andsignaling and preferably takes the form the of the Session InitiationProtocol (SIP).

As shown in FIG. 3, RTP is the preferred protocol for transportingreal-time data across the Internet. See H. Schulzrinne, S. Casner, R.Frederick and V. Jacobson, “RTP: A Transport Protocol for Real-TimeApplications,” Request for Comments (Proposed Standard, RFC 1889,Internet Engineering Task Force (January 1996) which is herebyincorporated by reference in its entirety. RTP is a “thin” protocolproviding support for applications with real-time properties, includingtiming reconstruction, loss detection, security and contentidentification. In addition, RTP provides support for real-timeconferencing for large groups within an intranet, including sourceidentification and support for gateways, such as for audio and videobridges, and multicast-to-unicast translators. RTP offersquality-of-service feedback from receivers to the multicast group aswell as support for the synchronization of different media streams.

In FIG. 3, the combined stack of the IP, UDP and RTP protocols 88, 90and 92 add 40 bytes to every packet for low-speed links and highlycompressed audio, and 20 bytes for 20 ms of 8 kb/sec. audio. Thus,header compression is desirable.

As noted above, the protocol stack 80 of FIG. 3 preferably employs theSession Initiation Protocol (SIP) for establishing multimedia exchangeswith one or more parties. Instead of using telephone numbers, SIP usesaddresses in the form user@domain or user@host. This address, forexample, can be identical to a person's e-mail address.

SIP provides the standard PBX or CLASS functionality, such as callforwarding, call waiting, caller M, call transfer, “camp-on,” “callpark,” and “call pickup.” “Camp-on” allows an attendant-originated orextended call to a busy single-line voice station to automatically waitat the called station until it becomes free while the attendant is freeto handle other calls. “Call park” allows a user to put a call on holdand then retrieve the call from another station within the system. “Callpickup” allows stations to answer calls to other extension numberswithin a user specified call pickup group. Many of these featuresactually require no signaling support at all, but can be implemented byend system software. SIP is designed as a variant of HTTP/1.1, whichallows easy reuse of HTTP security and authentication, content labelingand payment negotiation features.

SIP further employs a calendar-based call handler. The call-processingsoftware accesses a user's personal appointment calendar and answers thephone accordingly. The user can define categories of callers and preset,based on the calendar entry, whether and where their calls areforwarded. The information released to the caller if calls are notforwarded may range, for example, from “is currently not available” to“John Smith is in a meeting until 3 p.m. in Room 5621 with Jane Doe,”depending upon the caller's identity. The call handler can also beintegrated with a call processing language, a state-based scriptinglanguage that allows to construct voice-mail systems or automatic callhandling systems in a few lines of code. The call handler also managesthe translation between ISDN calls and Internet telephony calls.

FIG. 4 is a high-level hardware block diagram showing a preferredembodiment of an packet data network telephone 100 according to thepresent invention. As will become apparent throughout this disclosure,the device 100 is a relatively low cost interface product to place voiceand data onto a packet data network, such as Ethernet LAN's, intranetsand the Internet. Therefore, the device 100 will generally be referredto as a network appliance to reflect the broad applicability of thisstand alone device.

The network appliance 100 provides audio and video communications acrossa local area network (LAN), Internet or other Ethernet network, andgenerally includes: a network (e.g., Ethernet) controller subsystem 110;a digital signal processing subsystem 120; a signal conversion subsystem130; and a user interface subsystem 160 coupled to both the signalconversion subsystem 130 and the digital signal processing subsystem120. The telephone 100 further includes a power supply, ROM 142 and RAM152. The user interface subsystem may include a speaker 161, amicrophone 162 and other user controls 169 as discussed below and withreference to FIG. 5. Interface circuitry 135 for data acquisition andcontrol functions can also be coupled to the signal conversion subsystem130. Alternatively, such I/O circuitry can be directly coupled to DSP120.

The network controller subsystem 110 is interposed between the DSP 120and the external data network and as such provides and receives datapackets to and from the data (Ethernet) network. The Ethernet controllersubsystem 110 also instructs the digital processing subsystem 120 toaccept data received from or to provide data to the Ethernet network. Inaddition, the network controller subsystem can act as an initialgatekeeper by rejecting and discarding corrupted or unwanted datapackets received from the Ethernet network.

FIG. 5 is a block diagram which illustrates the present networkappliance in further detail. As shown in FIG. 5, a preferred embodimentof the network controller subsystem 110 includes an Ethernet controller112, a service filter 114 (10Base-T transformer) and at least one RJ-45socket 116. Among other things, the network controller subsystem 110performs the following functions: interfacing the network appliance tothe Ethernet network; sending and receiving Ethernet packets; informingthe DSP subsystem 120 to accept the data when the data is available fromthe Ethernet; receiving the packets from the DSP subsystem 120 andsending same to the Ethernet; and rejecting and discarding unwantedpackets from the Ethernet.

As shown in FIG. 5, the Ethernet Controller 112 is preferably theAM79C940 Media Access Controller for Ethernet (MACE) available fromAdvanced Micro Device (AMD). The MACE device is a slave register basedperipheral. All transfers to and from the system are performed usingsimple memory or I/O read and write commands. In conjunction with a userdefined DMA engine, the MACE chip provides an IEEE 802.3 interfacetailored to a specific application.

Individual transmit and receive FIFOs decrease system latency andsupport the following features: automatic retransmission with no FIFOreload; automatic receive stripping and transmit padding; automatic runtpacket rejection; automatic deletion of collision frames; direct FIFOread/write access for simple interface to DMA controllers or I/Oprocessors; arbitrary byte alignment and little/big/medium memoryinterface supported; and 5 MHZ-25 MHZ system clock speed.

Referring again to FIG. 5, the digital signal processing subsystem 120includes a digital signal processor (DSP) 122 and related logicalcircuits, which include a read-only memory (ROM) 142, a random accessmemory (RAM) 52, and a erasable programmable logic device (EPLD) 124.The digital signal processing subsystem 120 provides the followingfunctions: digital signal processing, such as speech compression; callprogress tone generation, and ring signal generation; general “glue”logic to interconnect DSP, memory and I/O devices; network protocolprocessing; call flow control and finite-state-machine implementation;keypad activity detection and decoding; and display control.

As shown in FIG. 5, the DSP 122 used in a preferred embodiment of thenetwork appliance can be any suitable commercially available DSP, suchas Texas Instruments' TMS320C32. The TMS320C32 DSP has the followingfeatures: parallel multiply and arithmetic logic unit (ALU) operationson integer or floating-point data in a single cycle; general-purposeregister file; program cache; dedicated auxiliary register arithmeticunits (ARAU); internal dual-access memories (512 double words); twodirect memory access (DMA) channels; one serial port; two timers; oneexternal memory port; and a multiple-interrupt structure.

In addition, the TMS320C32 DSP includes four external interrupts and sixinternal interrupt resources. The external interrupt can be triggereddirectly by the external pins. The internal interrupt can be triggeredby programming the individual peripherals, such as serial port, DMAcontroller, and timers. In addition, all these interrupt sources can beprogrammed as the DMA channel interrupt via CPU/DMA enable register, IE.The TMS320C32 DSP also includes a flexible boot loader which enables themain control program for the network appliance automatically loaded fromone of three different external memory spaces or the serial port,whichever is appropriate as determined by the activity of the externalinterrupts of INT0 to INT3 when the DSP 122 is initialized, such as atpowered on.

The DSP 122 is generally configured to include the following resourceassignments. External interrupts include: INT0: “System boot from0x1000” indication. when the system is powered on and into is active,the DSP will boot the program from external memory space 0x1000; INT1:DMA0 external interrupt signal, used for receiving packets from thenetwork controller 112; INT2: DMA1 external interrupt signal, used forsending packets to the network controller 112; INT3: AM79C940 packetstate and error message interrupt. A sample DSP memory map for use in anembodiment of the present network appliance is shown in FIG. 6.

Referring again to FIG. 5, the present network appliance has the userinterface subsystem 160 which includes: a key encoder 166, a liquidcrystal display (LCD) 164 and a hand set 163, which includes a keypad165, a microphone 162 and a speaker 161. The user interface subsystem160 components allow user interaction with the network appliance byproviding the following functions: user interface for input (keypad) andoutput (LCD); voice interface; ring alert output through speaker; andhandset or hands-free (microphone and speaker) communicationalternative. Through this interface 160 user commands are entered andaudio is sent and received to the user.

In addition, the LCD can have buttons adjacent to the display, such ason the side and below. The function of these buttons can operate as“soft keys” the function of which depends on the current state of thesystem. For example, when not answering calls, the display can shown aquick dial list and the time of day. In addition, after calls have goneunanswered or been forwarded to voice mail, the display shows can show alist of received calls. During the call, any other incoming calls aredisplayed, allowing the subscriber to switch between calls or bridge thecall into the existing call.

Alternatively, the user interface 160 of the present network appliance100 can be configured with a small touch screen (not shown) to replaceor supplement the LCD display and buttons. The touch screen, whichgraphically displays available functions and operations and responds touser contact on the display, provides an enhanced user interface, suchas for the entry of alphanumeric network addresses and other telephonyoperations.

FIG. 5 also shows the signal processing system 130, which includes PCMencoder and decoder that performs analog-to-digital (A/D) anddigital-to-analog (D/A) conversion, and an audio amplifier 134 coupledto the handset and the corresponding speaker 161 and microphone 162.Also provided is a power supply for providing positive and negative 5Vvoltage levels from a single AC or DC power supply adapter (“wallwart”). In the preferred embodiment of FIG. 5, negative voltage levelsare required by the LCD 164 and the PCM codec 132.

FIG. 7 is a block diagram which illustrates a memory interface 700suitable for use in the network appliance of FIG. 5. The memoryinterface 700 includes external memory modules 142 and 152, whichthemselves include 128 Kbyte of read-only memory (ROM) 142 for programstorage and at least 32 Kbytes of double word (32 bit) static randomaccess memory (RAM) 702, 704, 706 and 708. Due to the relatively slowspeed of the ROM 142, it is preferable that the network applianceinitializes the main program from the ROM and stores this program in therelatively fast RAM for run time execution.

FIG. 8 is a block diagram that shows an exemplary interface between theDSP 122 and the Ethernet controller 124 in accordance with a preferredembodiment of the present invention. The 32 registers of the Ethernetcontroller 124 are memory mapped at the 0x810000 memory space of the DSP122 as shown in FIG. 6. Preferably, the first two registers arereceiving and transmitting “first in, first out” (FIFO) queues. The DSP122 exchanges the data with the Ethernet controller 124 via a 16 bitdata bus 802.

FIG. 9 is a schematic diagram which illustrates an interface between theDSP 122 and the PCM codec 132 in accordance with a preferred embodimentof the present invention. As shown in FIG. 9, the DSP 122 connects tothe PCM codec 132 via an internal serial port 902. The serial port onthe DSP 122 is an independent bidirectional serial port.

As shown in FIG. 5, the DSP 122 is also operatively coupled to the LCD164. The LCD control interface is mapped at the DSP addresses shown inFIG. 10. In one embodiment of the present invention, the LCD 164 is a120×32 pixel LCD such as the MGLS-12032AD LCD, manufactured byVazitronics. Since the access speed of the LCD is generally slow, datadisplayed by the LCD can be mapped into the STRB0 (1X1000) memory spaceof the DSP 122, which is the same memory space as ROM memory space.Preferably, the LCD timing logic is the same as the timing logic for theDSP 122. However, when the LCD is composed of a left-half and aright-half, such as in the MGLS-12032, it is necessary to control andprogram for both of halves of the LCD when displaying an entire linemessage.

FIG. 11 is a block diagram showing the software architecture for thepresent network appliance. As shown in FIG. 11, the processingarchitecture for the present network appliance is generally organizedinto three levels: the ISR (Interrupt Service Routine) level 1110; theoperating system or Process level 1120; and the application or TaskLevel 1130. An exemplary list of functions and tasks which can beperformed at each of the software levels is provided in FIG. 13.

The lowest level, the ISR level 1110, includes interrupt handlers andI/O interface functions. The ISR level 1110 serves as the interfacebetween the process level 1120 and the network appliance hardware shownin FIGS. 4 and 5.

Above the ISR level 1110 is the process level 1120, or operating system,which is preferably a real-time multitasking micro-kernel, such asStarCom's CRTX Embedded Real-time micro-kernel. Generally, the processlevel software 1120 (micro-kernel) performs memory management, processand task management, and disk management functions. In a preferredembodiment of the present invention as shown in FIG. 12, themicro-kernel supports three scheduling mechanisms: a Real-time EventFlag Manager 1222; a Delayed Task Manager 1224; and a Scheduling Manager1226. The micro-kernel has three separate queues for the three differentmechanisms above, respectively.

The Real-time Event Flag Manager 1222 is used to trigger the executionof real-time events by way of setting flags. If a flag is set to an “ON”condition, the task associated with the flag is immediately executed.For example, an interrupt service routine would set a particular flagwhen a certain event occurred. Flag events are entered on a flag queuewith an associated task address.

The Delayed Task Manager 1224 is responsible for timed events. A timedtask, such as a fail-safe or “watchdog” task, can be executed after acertain time delay. If a certain event does not occur within a certaintime frame, the timer triggers the task causing it to be executed.Another example is the repeated execution of a task controlled by aperiodic timer. In an exemplary embodiment, there are 10 timer entries.Each timer is loaded with a tick count and is then decremented on everytimer tick from the hardware's interval timer. When the count reacheszero, the task associated with the timer is scheduled on the task queue.The Scheduling Manager 1226 scans the task schedule queue looking forscheduled tasks. Upon discovery of an entry in the queue, control ispassed to a scheduled task.

FIGS. 13 a-f are tables which list exemplary software tasks andfunctions which can be part of the task level software (FIGS. 13 a-c),process level software (FIG. 13 d) and ISR level software (FIGS. 13e-f). For the purposes of the present invention, the terms “task” and“function” as referred with respect to the software architecture areconsidered to be synonymous. However, “tasks” are generally executed bythe Scheduling Manager 1226, whereas “functions” are generally called bytasks or other functions. The application tasks, such as the callprocessing (Call_task) and IP processing (IP_Send_task and Ercv_task,etc.) tasks, are scheduled by the Process level software 1120. Theexecution of such tasks is a result of a prior scheduling by an ISR,another task, or by the current task itself.

FIGS. 13 A-F illustrate exemplary function and procedure definitionscalled in an event driven operation performed by the present packet datanetwork telephone software of FIG. 11. The functions, which are calledon the occurrence of various events, enable operation of the packet datanetwork telephone/system and include gross operations such as:initializing the Packet data network telephone/system; processing ARPdata; encoding voice data; processing message data; processing IP data;decoding voice data; transferring analog and digital data to and formcorresponding buffers; and performing “watchdog” functions.

Initialization of the packet data network telephone appliance includesthe steps of hardware initialization and task scheduling. After poweron, the DSP 122 will automatically transfer the main program from theROM 142 to the RAM 152 (boot operation). Hardware initialization occursin a traditional manner, including the steps of: initializing the stackpointer, external bus interface control register, global controlregister of the DSP, interrupt vector for the ISR, and the like.

After completion of the hardware initialization and preliminary taskscheduling, processing control is returned to the process level(micro-kernel) 1120. The CRTX micro kernel 1120 and the scheduled tasksthen control further processing.

Referring again to FIG. 13 a, the task level software of the presentnetwork appliance includes Address Resolution Protocol (ARP) processing.ARP is a known TCP/IP protocol used to convert an IP address into aphysical address (called a Data Link Control (DLC) address), such as anEthernet address. A host computer wishing to obtain a physical addressbroadcasts an ARP request onto the TCP/IP network. The host computer onthe network that has the IP address in the request then replies with itsphysical hardware address.

FIG. 14 is a flow diagram illustrating of an ARP request outputprocedure 1400, ARP_Out( ). As illustrated in FIG. 13 B, ARP_Out( ) is acomponent of the task level software which receives an IP address to beresolved, and outputs a corresponding MAC address. When a ARP requestbegins (step 1402) the ARP_Out( ) function first checks the requested IPaddress from a local ARP cache table, arptable (step 1404). If thecorresponding entry is RESOLVED at step 1406, then ARP_Out( ) copies theMAC address from arptable to the requested parameter and returns a ARPOKstatus flag (step 1408). Otherwise, the procedure will allocate an entryin the arptable and schedules a ARP request (step 1410). As furthershown by step 1410, a MAC address, i.e., “handle,” of the arptable isreturned to the main program (c_int00( )). According to the handle, thesoftware then checks the corresponding entry's ae_state.

FIG. 15 is a flow diagram of an exemplary ARP request input procedure1500, ARP_In_task( ), which is a component of the task level softwarelisted in FIG. 13 A. The ARP_In_task receives an ARP packet, and eithermodifies the arptable or queues an ARP reply if the incoming packet isan ARP request. When receiving an ARP packet (step 1502) the softwarewill check whether the packet's ARP hardware or protocol types match(step 1504). If the types do not match, control is returned to the mainprogram (step 1506). If one or both of the types match, then thesoftware checks if the destination host is the present host (step 1510).If the destination host is not the present host, then control isreturned to the main program (step 1508).

As further shown in FIG. 15, if the destination host is the currenthost, then the ARP_In_task next checks the ARP table to determinewhether there is a corresponding ARP entry for the incoming packet (step1512). If an entry is found (step 1514), then the new MAC address iscopied into the existing entry and modifies the entry's “Time to Live”(TTL) to a new value (step 1516). A TTL is understood by those withskilled in the art to be a field in the Internet Protocol (IP) thatspecifies how many more hops a packet can travel before being discardedor returned to the sender. However, if there is no such MAC entry isfound in accordance with step 1513, then the ARP_In_task adds a new MACentry in the ARP table (step 1518). If the MAC entry is in a PENDINGstate (step 1520), it is then changed to a RESOLVED state and the MACaddress is copied to the target entry (step 1522). If the incoming ARPpacket is an ARP request from another host, an ARP reply packet is sentby queuing the IP_Send_task, steps 1524 and 1526. Control is thenreturned to the main program (step 1528).

In addition to the ARP input and output processes, ARP processing at thetask level includes an ARPTimer_task( ), which is a delayed loop taskused to maintain the ARP entry table arpentry. Nominally, theARPTimer_task( ) is generated once per second. The main purpose of theARPTimer_task( ) is to decrease the “Time to Live” (TTL) of the ARPentry and to resend the ARP request during the pending state in case theprevious ARP request is lost.

Task level processing can also include processing operations associatedwith the coding and decoding of audio packets. The Codec_task generallyincludes a SpeechEncode( ) function, which encodes speech data from theADBuf buffer to the EncodeBuf according to the algorithm indicated by“type” parameter. The coded data is then sent out via the queueIP_Send_task, with the “RTP” parameter set.

Task level operations can also include Internet protocol (IP)processing. The general IP processing operations are illustrated in theblock diagram of FIG. 16. As shown in FIG. 16, IP processing includesthe steps of: transmitting and receiving Ethernet packets, step 1602;multiplexing and de-multiplexing IP packets, step 1604; and packetizingand de-packetizing Ethernet, Internet Protcol (IP), User DatagramProtocol (UDP), Real-Time Transport Protocol (RTP) and AddressResolution Protocol (ARP) packets, step 1606.

In accordance with step 1602 of FIG. 16, Ethernet packet transmissioncan be performed using direct memory access (DMA) channels of theEthernet controller 112. DMA is a technique for transferring data frommain memory to a device without passing it through the CPU. Since DMAchannels can transfer data to and from devices much more quickly thanwith conventional means, use of DMA channels are especially useful inreal-time applications, such as the present network telephony system.

The network controller 110 preferably supports a plurality of DMAchannels, such as the DMA1 channel of the Ethernet controller 112 thatcan be used for packet transmission. When an Ethernet packet is readyfor transmission, the DMA1( ) function, an ISR level function, is calledby setting the source address (Ethernet packet buffer, ESend),destination address (Ethernet controller's transmit FIFO), and a counter(the packet length). Examples of Ethernet transmit data structures areprovided in FIG. 17. The DAM1( ) function then starts the DMA1 channel.When the counter reaches zero, the DMA1 stops and waits for the nextcall.

FIG. 18 is a block diagram which shows the data flow between an audioinput buffer 1802, a UDP buffer 1804 and ARP table 1806 to the Ethernetinterface (Ethernet Transmit FIFO) of the Ethernet network controller112. As further shown in FIG. 18, data from the audio input buffer 1802,the UDP buffer 1804 and the ARP table 1806 is sent to an IP output queue1810, and is arranged to indicate the protocol type, source pointer anddata length. Instead of queuing the sending data, the IP_Send_task isqueued by process level software (micro-kernal) 1120. The protocol typessupported by the IP_Send_task generally include UDP, RTP, ARP_REQUEST,ARP_REPLY. IP_Send_task is used for packet transmission and Ethernetpacketizing. Preferably, IP_Send_task is scheduled by other tasks orfunctions such as SIP_task, ARP_Out( ), SpeechEncode( ), etc. Once theIP_Send_task is run, it checks the protocol type of the data. This taskthen encapsulates the output data into the corresponding Ethernet packetin the ESend buffer. Finally, the packet is sent out via the assignedDMA channel (DMA1).

FIG. 19 is a data flowchart further illustrating packet receiving andde-multiplexing operations. The de-multiplexing is realized byscheduling different tasks for different protocols in the Ercv_task. Infurther accordance with step 1602 of FIG. 16, Ethernet packets arereceived in the receive data FIFO memory (step 1902) and are furtherprocessed by a DMA0 channel controller (step 1904). Since the DSP 122doesn't know when the packets will arrive, the DMA0 channel is activeall the time (i.e., it does not stop even the counter reaches zero).When a packet arrives, the DMA0 channel will automatically copy it fromthe Ethernet controller's receiving FIFO to the Ethernet receivingbuffer, Ercv (step 1906). The DMA0 channel stops when there is no dataavailable in the FIFO.

ERcv_task is a flag trigger task for Ethernet packet de-packetizing andIP packet de-multiplexing (step 1908). The Ercv_task functions asfollows: first, a PacketCheck( ) function is called to check theincoming packet. The PacketCheck( ) will return the protocol type of thepacket, or NULL if the packet is invalid. Second, depending on thereturned protocol type, the ERcv_task will trigger the different tasksto process the received packet, RTP_In_task for “RTP” packet (step1910), ARP_In_task for “ARP” packet (step 1912) or UDP processing tasks(step 1912) for UDP packets, for example.

Referring to FIG. 13 C, SpeechDecode( ) is a voice decoding functionassociated with the RTP processing of step 1910. First, a SpeechDecode() task checks if there are data available in the decoding buffer,DecodeBuf. If data is available, e.g., RcvFlag is SET, thenSpeechDecode( ) decodes it according to the data type of receiving data,PCM (G.711), G.723, G.729, for example. The decoded data is sent intothe D/A buffer, DABuf.

The A/D and D/A interrupt routine can be triggered by an internalinterrupt source, e.g., Rint0( ). Preferably, the A/D and D/A interruptroutine is triggered by an 8 kHz sampling frequency provided by the DSP.Since this routine is called frequently, Rint0( ) is preferably writtenin assembly language. The steps performed by Rint0( ) include the stepsof: reading a D/A sample from D/A buffer, DABuf; sending the sample tothe serial D/A port; obtaining a sample from the serial A/D port; savingthe A/D sample to an A/D buffer, ADBuf; and incrementing A/D and D/Abuffer pointers, ADPnt and DAPnt, by one.

FIGS. 20A and 20B are block diagrams which show an A/D and D/A“ping-pong” buffer scheme used by the software of the present invention.Further, if the current A/D pointer value (ADPnt) exceeds apredetermined buffer threshold (ADTh) then a flag is set in the flagtask queue indicating that service is required.

The A/D and D/A buffers can be divided into two parts, the upper buffer2002 a and lower buffer 2002 b, respectively. Both buffers can bedesigned as circular buffers. In this way, when the current pointerreaches the buffer bottom, it wraps around to its beginning. However,from the encoder and decoder point of view, it is used as a two-frameping-pong buffer (defined as upper frame and lower frame) scheme. Theoperation of this process is shown in FIGS. 20A and 20B. For A/Dconversion, when the upper (or lower) is full, the data in the upper (orlower) buffer will be passed through ping pong switch 2004 and copied tothe speech encode buffer, EncodeBuf 2006. For D/A conversion, if theupper (or lower) buffer is completed, a new frame of data will be copiedfrom the speech decode buffer, DecodeBuf, 2010 to the upper 2008 a (orlower 2008 b) buffer. This mechanism ensures that while the encoding (ordecoding) algorithm reads(writes) from one part of the buffer, the A/D(or D/A) sampling ISR can write (read) the other part of the bufferwithout conflict.

FIG. 21 is a state transition diagram of a Call_task subroutine used inan exemplary embodiment of the present network appliance. Call_task is alooped task which handles the call procedure. As shown in FIG. 21, the“Idle” state 2102 occurs when there is no call being made and there isno incoming call. When this condition exists, the Call_task loops in the“Idle” state 2102. The “DialTone” state 2104 exists when the receiverstate is OFFHOOK, or the handset state indicates HANDSFREE, and thus theCall_task state will change from the “Idle” state 2102 to the “DialTone”state 2104 when a OFFHOOK or HANDSFREE condition exists. These statesare generally entered by an input by a user through the user controls160 indicating that a call is to be initiated. When the Call_task stateis in the DialTone” state 2104, the Codec_task will be configured as“ToneMode, DialTone” and a dial tone is sent to the handset componentsof the user interface 160.

Referring again to FIG. 21, while in the “DialTone” state 2104, if anydigit key (‘0 . . . ‘9’, ‘*’ and ‘#’) or the redial button is pressed,the call state changes from the “DialTone” state 2104 to the “GetDigit”state 2106. In the “GetDigit” state 2106, the dial tone is stopped atthe handset.

After the callee's number has been input and an ENTER button has beenpressed by the user to indicate that dialing is complete, the Call_taskwill check if the input is valid. If the number is valid, a call entryis created by a function CreateSipCall( ) and the Call_task will go intoa “SIP” state 2108. Otherwise, if the input number is invalid, thenumber is requested again and the state remains at the “GetDigit” state2106.

While waiting for SIP_task processing, several decisions may be madedepending on the “SIP” state 2108. The “SIP” state 2108 is a globalvariable, SIP status, which is modified by the SIP_task according to itsstate transition. If the “SIP” state 2108 changes into SIP_Ring, theCall_task will change to the “RingBack” state 2114 and the Codec_taskwill be configured as “ToneMode, RingBack” mode. When the Codec_task isin the “ToneMode, RingBack” mode, a ring back tone is sent to thehandset.

From the “SIP” state 2108, if the “SIP” state 2108 changes to SIP_busy,the Call_task and thus the call will change into “BusyTone” state 2120and the busy tone will be played at the handset. It the “SIP” state 2108changes to SIP_Refused, appropriate messages will be displayed on theLCD screen related to the SIP_Refused state.

From the “RingBack” state 2118, if the “SIP” state becomesSIP_Connected, the Call_task state changes to the “Talk” state 2116.When the Call_task state is in the “Talk” state 2116, the Codec_taskwill configured as SpeechEncode and SpeechDecode mode.

For incoming calls, while in the “Idle” state 2102, if the “SIP” state2108 is SIP_Invite, the Call_task state changes to the “Ring” state 2114and the Codec_task will be configured as “ToneMode, RingTone.” When theCodec_task is configured as “ToneMode, RingTone,” a ring tone will beplayed on the loudspeaker. After the SIP state becomes SIP_Connected,the Call_task state will change into the “Talk” state 2116. Otherwise,if the SIP state becomes SIP_Cancel, which happens if the caller givesup the call, the Call_task state returns to the “Idle” state 2108.

While at the “Idle” state 2102, if the ENTER button is depressed, theCall_task calls the Setting_task. When the parameter setting program isfinished, it will return to Call_task.

During Call_task execution, if the hook state indicates the receiver isONHOOK, or a system error is found, the Call_task changes to the “Idle”state 2102, regardless of what the previous state is (except the “Ring”state 2114).

In the preferred embodiment of the network appliance as shown in FIG. 5,the key pad of the telephone has 17 keys for providing user inputs andcommands. The telephone key pad includes 10 digit keys, two special keysand five function keys are defined as shown in FIG. 22.

The Key_task is a loop delayed task which runs periodically, such asevery 0.1 seconds. When started, Key_task first calls the key( )function. If the return value is not “−1”, it means a key has beenpressed. Then, the KeyMap( ) function maps the input binary key word tothe ASCII key word. The Key_task then sets the corresponding member ofthe FuncKey structure. If the system is ready to accept the key input(the KeyRegEnable is indicated), the input key word is stored into theKeyBuf.

In addition, Key_task preferably supports four different input modes:digit input mode, IP address input mode, alphabet input mode, and listaddress input mode. Switching among the four modes can be done bypressing the ENTER button before dialing any number or alphabet when thehandset is picked up and a dial tone is heard. After input is completeand the ENTER button is pressed, the input numbers will be transferredto the current task (Call_task or Setting_task) by a message pipe. Ifthe Redial key is pushed, the task will copy the previous input from thebackup buffer, KeyBackup, to the KeyBuf. Then the data will betransferred to Call_task.

The operating system of the present network appliance preferablysupports a delayed task schedule scheme. The delayed task is similar tothe sleep( ) function in UNIX. However, a delayed task can also be apersistent task execution from a periodic timer when the task'srepeating flag is set. For delayed tasks, the process level software1120 requires an interval timer to provide a system tick. The system ofFIG. 5 uses the TMS320C32's timer1, TCLK1, as the system timer base.

The Clock_task is a looped delayed task which performs real time clockand calendar functions. It serves as the general clock to calculate anddisplay the current time, including the hour, minute and second. When acall is connected, it can display the call duration. When the phone ison hook, current year, month and date can also be displayed on the LCD.

Referring again to FIG. 11, the Network telephone software of thepresent invention includes several low-level functions that are includedas part of the software ISR level. Some of these low-level functions areI/O related functions, which are used with the telephone's 8-bit I/Oparallel port defined in FIG. 24. The low-level, I/O related functionsinclude: the “Hook” state monitor, Hookst( ); the Key input availabilitycheck and read, Key( ); handset and hands-free control, HandSet( );Ethernet controller reset, ENET_reset( ); volume control, AmpControl( );and software reset of the system.

The audio interface chip 136, which preferably takes the form of anLM4830, can be used to control switching between the handset and thehands-free mode. For example, the HandSet( ) function can write a ‘0’ tothe I/O port when “hands-free” mode is required or write a ‘1’ to theappropriate port when “handset” mode is required.

The low-level functions of the present invention also include theEthernet controller interrupt ISR, c_int03( ). The global messagestructure for use with c_int03( ) is defined for the state of theEthernet controller as shown in FIG. 25. Whenever a packet has beensent, or a received packet is complete, the Ethernet controller willinterrupt the DSP 122 to indicate the interrupt. The DSP 122 will readthe transmission and receiving states from the Ethernet controller'sregister and then store the state into the above state structure. Thisinformation can be checked by other tasks. In addition, these messagesare read after each packet transmission. Otherwise, the Ethernetcontroller will be blocked.

As noted above, it is preferred that the present network appliance ofthe present invention uses the RTP protocol to transmit and receivespeech packets in real time. The RTP packet is encapsulated in an UDPpacket. The IP_Send_task and the RTP_In_task modules operate to createand parse RTP packets. FIG. 26 shows an RTP header structure for RTPpacket processing.

When the IP_Send_task gets a request to send a RTP packet, it firstgenerates an Ethernet and UDP header. Next, it adds the RTP header inthe Ethernet packet transmission buffer. Finally, the RTP data is copiedinto the RTP data area and is sent over the data network.

FIG. 27 shows a data structure for use with a tone generation function,Tone_task( ). The parameters described in FIG. 27 are illustrated in thetone generation timing diagram of FIG. 28.

Tone_task is a delayed task which can be executed about every 0.1second. It is used to count the tone active and stop duration defined inthe ToneType structure. Tone_task sets ToneState to ACTIVE during burstand STOP during silence. Different active and stop duration generatesdifferent tones. They are: Dial tone, continuous tone (no stop); Busytone, burst 0.5 s and silence 0.5 sec; Ring back tone, burst 2 sec andsilence 4 sec; Ring signal, burst 0.8 sec twice in two seconds, thensilence 4 sec.

Preferably, a ToneGenerate( ) module generates a one frame 400 Hz toneor a 2400 Hz ring signal defined by “mode” parameter when ToneState isACTIVE. Otherwise, one frame silence signal is provided.

The network appliance of the present invention uses UDP as its transportprotocol for SIP. SIP_task is a looped task that handles SIP signaling.Since the present network appliance can be used either as a caller or asa callee, SIP_task operates both as a UAC (User Agent Client) and a UAS(User Agent Server).

FIG. 29 is source code which shows data structures used for processingthe SIP requests or responses in accordance with the SIP protocol.Tstate is the state transition structure used in SIP_In_task andSIP_task for SIP state transition. Parsed SIP messages are in the datastructure message_t. The structure call is defined for each call and thetotal call entries are defined by msg[MaxSipEntry].

FIG. 30 shows a state transition diagram of the SIP_task operating as aclient (e.g., a caller). When the SIP phone starts a call, it works as aclient. A call will be created via the following steps: a call entrymsg[CurrentIndex] is allocated when the phone is picked up and the flagof the call is SET; CreateSipCall( ) creates a SIP packet according tocurrent setting and dial inputs, wherein the SIP package is used as thereference of the call and the us_state is set to UAC; SIPParse( )generates the message structure (msg[CurrentIndex].m) for the call fromabove packet; the SIP_task will check if there are any active calls—ifthere is a call (msg[i].flag is SET), SIP_task will create thecorresponding request according to the SIP specification and the SIPstates will be updated in SIP_task as shown in FIG. 30.

FIG. 30 shows an exemplary state diagram for client (caller) operations,referred to as a UAC state transition diagram of SIP_task. From anInitial state (step 3002) a Calling state is entered and a SIP_taskretransmits a SIP INVITE request periodically (T1) until a response isreceived (step 3004). Nominally, T1 is 500 ms initially and doublesafter each packet transmission. (Step 3006) T2 is nominally 32 seconds.If the client receives no response, SIP_task ceases retransmission whenT2 timer expires and SIP state will be changed to Cancel (step 3008). Ifthe response is provisional, the client continues to retransmit therequest up to seven times. When a final response is received, the statewill change to Completed and a ACK will be generated (step 3010). Whenthe caller gives up, the state will changed to Bye state (step 3012).BYE requests are also retransmitted during the interval of T1 until T2expires for the purpose of reliable transmission. The variable,SIP_Status, will be changed according to the response received as shownin FIG. 31. For example, if a 3xx response is received, SIP_task willinitiate another call to the redirected address. Other final responsescan be displayed on the LCD.

When the network appliance receives a call, the SIP_task functions as aSIP UAS (server). The incoming packets are processed as follows:UDP_In_task accepts the incoming UDP packet and sends the packets toSIP_In_task along with its source IP address and port number.SIP_In_task processes the packet according to the SIP specification andupdates the states accordingly. SIP_task will monitor the receiverstate, set and decrease the T1 and T2 timer of each call and update theSIP states if necessary.

FIG. 32 illustrates an exemplary state transition diagram of a SIP UAS.While the SIP_task remains at an Initial state (step 3205), it listensto the incoming SIP packets. If an INVITE request is received, itgenerates a Ringing(180) response and its state changes to Invite andthe Sip task module would advance to a Proceeding step (step 3210). If acalled party picks up the telephone, the status changes to Picks Up andthe process advances to Success (step 3215), indicating a successfulcall session has been established. If the called party does not pick up,the status changes to Failure and the process advances to the Failurestate (step 3220). After success or failure, the client will acknowledgethe current status and advance the process to the Confirmed state (step3225). When the calling party terminates the session, the status changesto Onhook and the process advances to Bye (step 3230) indicating thatthe current session has been completed.

As set forth herein, the network appliance is a stand alone devicecapable of initiating and receiving telephone calls on a packet datanetwork. While the stand alone architecture described herein offers manyattendant advantages, such as its relatively low cost to implement,similar software architecture and functional definitions described inconnection with the stand alone appliance 100 can also be provided on aPC based telephone device. In such a case, a conventional personalcomputer having a microphone, speakers and suitable network interfacecard, is provided with software to operate consistently with the mannerdescribed above. Of course, obvious changes are effected in thisembodiment, such as the user interface components and functions beingperformed by conventional elements of the PC, e.g., the keyboard,monitor, mouse and the like. A GUI interface to the telephonefunctionality is provided by the software to enable the desiredtelephony functions.

The network appliance of the present invention, in addition toperforming traditional telephony functions, can also provide a costeffective interface between the network and the environment. Whileequipping sensors with Ethernet interfaces is not feasible, due to thelarge number of ports required and the cost of the minimal hardwarerequired, the network appliance of the present invention can become thegathering point for a number of digital and analog sensors. This isaccomplished generally by coupling the external sensor to the networkappliance via the conventional I/O circuitry 135 which is coupled to theDSP 122. The I/O circuitry can take the form of simple buffers, A/Dconverters, registers and the like. This feature is particularly usefulin environments that have phones for security reasons, e.g., elevators,lobbies, shop floors, garages, etc. Examples include: Passive infrared(PIR) digital sensor for detecting the presence of people—this can beused for automatically forwarding calls if nobody is in the office or aspart of a security or energy management system; analog or digital lightsensor to detect whether the office is occupied; analog temperaturesensor; smoke, carbon monoxide and radiation detectors; and contactclosures for security systems. Thus, the present network applianceprovides a point of system integration.

To provide further enhanced I/O capability, the I/O circuitry can becompatible with local control protocols such as the X10 and CEbusprotocols which are recognized standards for controlling line-powereddevices such as lighting or appliances. Adding such and interface to thephone provides for network-based control of such devices.

FIG. 33 illustrates a system employing the present network appliance forestablishing calls between two or more parties on the network. Thesystem generally includes one or more stand alone network appliances100, such as described above. In addition, the system can also includePC based telephony devices 3320, such as a network enabled PC operatingsuitable network telephony software which is protocol compliant with thenetwork appliance 100. Each telephony endpoint can be referred to as anode and has a specific SIP address. By employing this specific address,any node acting as a calling party (client) can directly initiate a callsession with any other node on the network (server).

The system preferably also includes a redirect server 3325 which can beaccessed by the various nodes on the network to provide enhancedservices, such as a directory service, call forwarding, call branching,call messaging and the like. For example, a calling party wishing toinitiate a call to JOHN SMITH can enter the SIP address for that personif it is known, such as sip:john.smith@work.com. If, on the other hand,the calling party does not know the SIP address of the party, thecalling party can contact the redirect server 3325 with a request tobegin a session with JOHN SMITH. The redirect server includes databaseswith registration information for various parties and can return the SIPaddress to the calling party or forward the call request to the properSIP address. In addition, the called party may have multiple sipaddresses such as john.smith@home, john.smith@office, john.smith@lab andthe like. The redirect server can provide a session initiation signal toeach of these addresses and establish a connection between the callingparty and the first contacted node that responds to the initiationrequest. Similarly, parties can periodically register with the redirectserver to indicate the current SIP address where they can be contacted(call forwarding feature).

The network appliance 3305 can be configured to interface to one or moresensors 3310. Signals from the sensors are received by the networkappliance 3305 and can be sent along the network to a desired networknode. The signals from the sensors can be detected periodically by atimer in the network appliance and sent to a SIP address stored inmemory. Alternatively, the sensor signals can be measured by the networkappliance 100 based on a command received from another node (polled by aremote network node) or can be measured based on a received interruptsignal indicating a change of state of the sensor (interrupt driven).For example, the network appliance 100 can be used as a security systemcommunication device which reports the status of various security sensorpoints to a central monitoring station. In such a case, the appliancecan periodically check the status of the connected sensors, such as doorsensors, fire sensors, passive infrared detectors and the like, andreport to a central station node the current status. In the event of astatus change that would indicate an alarm condition, the appliance 100could generate a call session with the central station and report thiscondition as well. Of course, the same appliance which is acting as analarm communicator can also provide full telephony functions as well. Inaddition, while a simple security application was described, it willalso be appreciated that various other data collection and controlapplications generally known as SCADA (site control and dataacquisition), can be implemented using the present network appliance100.

To maintain lifeline service during power outages, the network applianceof the present invention can be equipped with a rechargeable battery,possibly integrated into a wall transformer.

As many locations are currently equipped with only one Ethernetinterface, the network appliance of the present invention should providea two-port Ethernet hub, with an external RJ-45 interface. This providesfor simultaneous operation of both the telephony device and networkenabled computer.

In addition to audio data, the present network appliance can alsoreceive and transport video data. For example, a video input interface,either analog or through a USB (Universal Serial Bus) can be operativelycoupled to DSP 122 to implement this feature.

The present network appliance 100 can also be coupled to a suitablewireless Ethernet interface to allow the equivalent of a cordless phone.

The following protocols can be added to the present network appliance100 to provide expanded functionality: DHCP and RARP for automaticassignment of IP addresses; IGMP for subscribing to multicast groups;RTSP for retrieving voice mail and distinctive ringing signals; SAP forlistening to announcements of multicast “radio” events; and DNS for nameresolution (subject to available program memory space).

In addition to basic telephony operations, the present network appliancecan also provide high level telephony functions. For example a “Do notdisturb” feature can be provided that automatically forwards calls for agiven duration to a designated location as specified by a SIP addressinput by the user. Each time the feature is selected, such as bydepressing a button on the user interface, the time increases by apredetermined interval (e.g., 15 minutes).

“Call logging” can also be provided wherein the SIP address and relatedinformation regarding incoming calls is logged by storing theinformation in memory, with the ability to call back the calling partyby scrolling through the list and selecting the SIP address of thecaller from the log by user interaction via the user interface subsystem160.

The network appliance can also include an “Automatic address book.”Through user input or via a server connected on the network, the networkappliance can acquire a speed dial list or a list of names stored in itslocal memory which a user can scroll through (using the SIP “multiplechoices” response);

An “Interface to voice mail system” feature can display all unansweredcalls that have come in, including the time of call, the caller, thesubject and urgency of the call and whether the caller left voice mail.Calls can be ordered chronologically or by urgency. The call displaypreferably features five soft buttons: to delete the entry, to moveforward and back through the list, to return the call and to retrievethe message.

“Distinctive ringing” is a feature wherein the appliance 100 isprogrammed to announce certain callers by a distinct sound clip, such asa distinctive ring, melody or the name of the caller. In this case asmall database associates a caller, or a class of callers (e.g., friend,customer, urgent) to a particular selected ring response. The sound clipis played either from memory or retrieved from a server;

“Call forwarding” is a further feature which can be implemented in theappliance 100. Typically, calls are forwarded by the proxy redirectserver. However, the network appliance 100 can also perform simpleforwarding itself, as described above for the “do not disturb” button.The redirection may take the form of calling the phone from anotherphone with a REGISTER command, to implement follow-me calls. Also,automatic forwarding of calls from certain domains or during certainhours is readily implemented without use of a redirect server.

“Intercom” mode is a feature where incoming calls are “picked up”automatically, with the microphone disabled until a push-to-talk buttonis pressed or the receiver is lifted. This can also be used as part of asecurity public address system.

“Baby monitoring” features allow the network appliance to act as aremote audio monitoring device. For example, on receipt of an incomingcall, the network appliance 100 is activated with the speaker disabledbut with the microphone automatically enabled such that the callingparty can listen to the environment where the called appliance islocated. This feature can be selectively engaged, such as by apredetermined code or caller identity;

An “Internet radio” feature allows the network appliance 100 toautomatically play radio stations supplied by a local RTP multicastserver or other streaming media source when a call is not being receivedor initiated. The appliance 100 can listen for SAP announcements and candisplay the station list on the display, with soft buttons. Any incomingphone call interrupts the current radio program.

The present network appliance can also maintain a “Callee list.” If aprevious call was successful, the callee's address is automaticallyentered into a portion of memory used as a local guide-dial list. Whenthis party is to be dialed again, the callee can be selected by theupward or downward key from the callee's list. This is generally a FIFOtype memory structure which automatically purges old entries andreplaces them with more current entries; and

“Redial,” which allows single key dialing of either the last numberdialed or the last callee.

In addition, “Speech processing enhancement,” such as silencesuppression, comfort noise generation, and echo cancellation can also beincluded in the present network appliance in a manner which is wellknown in the telephony art.

Thus, a network-based telephone that is a stand-alone “Internetappliance” that allows the user to make phone calls within a local areanetwork (LAN) or across the Internet has been disclosed. Its core is asingle digital signal processor (DSP) (a microcontroller optimized forprocessing audio and video data). It provides services that are asuperset of those of a regular telephone, but connects and Ethernet datanetwork instead of to the PSTN (Public Switching Telephone Network).Since Ethernet running at 10 Mb/s can use the same twisted-pair wiringused for analog and digital phones, the Packet data network telephonedoes not require rewiring customer premises. A minimal system consistsof two Packet data network telephones connected by an Ethernetcross-over cable. A multi-line basic PBX can be implemented consistingof any number of Packet data network telephone connected to an Ethernethub or switch. This “PBX” can scale to any number of phones, simply byadding Ethernet capacity and ports. The Packet data network telephoneshares the Ethernet with other LAN services. In almost all cases, voicetraffic will be a small fraction of the network capacity. (A singlevoice call consumes about 16 kb/s of the 10 Mb/s capacity.) The Packetdata network telephone offers voice communications, implementing thecustomary features of PBXs. However, the present network appliance mayuse a server located in the LAN or the Internet to provide additionalfunctionality, such as user location and directory services, callforwarding, voice mail, attendant services.

A PBX based on the current network appliance can reach traditionalphones through an Internet Telephony Gateway (ITG). Such a gatewayconnects to the PSTN using either analog lines, ISDN basic or primaryrate interfaces or digital trunks (such as T1/E1). ITGs have recentlybeen introduced as commercial products, with capacities of one to about240 lines.

Although the present invention has been described in connection withparticular embodiments thereof, it is to be understood that variousmodifications, alterations and adaptions may be made by those skilled inthe art without departing from the spirit and scope of the invention. Itis intended that the invention be limited only by the appended claims.

1. A network appliance for providing packetized data over a packet datanetwork, comprising: a network controller subsystem coupled to saidpacket data network; a digital signal processing subsystem coupled tosaid network controller subsystem, the digital signal processingsubsystem further comprising a computer program for detecting incomingcalls and initiating call sessions; a signal conversion subsystemcoupled to said digital signal processing subsystem; and a userinterface subsystem coupled to both the signal conversion subsystem andsaid digital signal processing subsystem.
 2. The network applianceaccording to claim 1, wherein said digital signal processing subsystemcomprises a digital signal processor (DSP) and one or more memorydevices coupled to said digital signal processor.
 3. The networkappliance according to claim 2, wherein said computer program implementsthe Session Initiation Protocol for detecting and initiating callsessions and performing call session control.
 4. The network applianceaccording to claim 3, wherein a unique SIP address is associated withthe appliance, said address being stored in at least one of said memorydevices.
 5. The network appliance according to claim 4, wherein thepacketized data includes audio data and wherein the user interfacesubsystem comprises: a handset comprising an input device, a microphoneand a speaker; and a display device.
 6. The network appliance accordingto claim 5, wherein said computer program implements a monitor feature,wherein on detection of a call directed to the appliance from a caller,a call session is automatically initiated with said microphone enabledand said speaker disabled during the call session.
 7. The networkappliance of claim 6, wherein identifying criteria of at least oneapproved caller is stored in at least one of said memory devices andwherein said digital signal processor receives identifying criteria fromthe caller and activates the monitor feature only if the receivedidentifying criteria matches at least one of the stored identifyingcriteria of said at least one predetermined approved caller.
 8. Thenetwork appliance of claim 7, wherein said identifying criteria areselected from the group including name, SIP address and password.
 9. Thenetwork appliance according to claim 3, wherein the computer programimplements a call forwarding feature, wherein at least one forwardingSIP address is stored in at least one of said memory devices, at leastone of said forwarding SIP addresses being selectable by a user via saiduser interface subsystem, and wherein on detection of a call directed tothe appliance from a caller, said call is redirected to the selectedforwarding SIP address.
 10. The network appliance according to claim 9,wherein said call forwarding feature is activated for a predeterminedtime in response to an input from the user.
 11. The network applianceaccording to claim 9, further comprising a sensor coupled to saidappliance for detecting the absence of a human being, wherein said callforwarding feature is activated in response to a signal from saidsensor.
 12. The network appliance according to claim 1, wherein the userinterface subsystem includes an output device and wherein the computerprogram implements a streaming media mode wherein streaming data isreceived from the network and is converted to perceptable signalsprovided to said output device.
 13. The network appliance according toclaim 12 wherein the output device includes a speaker and wherein whenno call session is in progress streaming data is received from thenetwork and is converted to audio signals provided to said speaker. 14.The network appliance according to claim 13 wherein the program revertsout of streaming media mode in the event a new call session isinitiated.
 15. The network appliance according to claim 12 wherein theoutput device includes a speaker and wherein streaming data isselectively received from the network and is converted to audio signalsprovided to said speaker.
 16. The network appliance according to claim12 wherein the streaming data is received from the network and isselectively forwarded to another device during a call session where thedata is convertable to perceptable signals by said device.
 17. Thenetwork appliance according to claim 12 wherein the output deviceincludes a video display and wherein streaming data includes streamingvideo data which is selectively received from the network and isconverted to video signals provided to said display.
 18. The networkappliance of claim 3, wherein the user interface subsystem includes adisplay device and wherein the digital signal processor detects the SIPaddress of callers and stores a plurality of caller SIP addresses in atleast one of said memory devices, said plurality of caller SIP addressesbeing displayable on said display device and selectable in response toan input from the user interface subsystem.
 19. The network appliance ofclaim 3, wherein the user interface subsystem includes a display deviceand wherein the digital signal processor stores a plurality of calledSIP addresses in said memory device, said called SIP addressescorresponding to the address of successfully initiated call sessions andbeing displayable on said display device and selectable in response toan input from the user interface subsystem.
 20. The network applianceaccording to claim 1, wherein said network controller subsystemcomprises an Ethernet controller and a service filter.